login | register
Wed 01 of Aug, 2007 [09:17 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Last Changes
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.44s
  • Memory usage: 2.23MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 3.15

Asterisk codecs

Asterisk Codecs


Asterisk supports the following codecs


To tell which codec is being used for a specific call:
sip show channels
iax2 show channels

To use with allow and disallow, here is the association table:

G.711 ulaw = ulaw (US standard)
G.711 alaw = alaw (European standard)
G.723.1 = g723.1 (pass-thru only)
G.726 = g726
G.729 = g729
GSM = gsm
iLBC = ilbc
LPC10 = lpc10
Speex = speex
ADPCM = adpcm

A typical use might be:
disallow=all
allow=alaw
allow=ulaw


File name extensions

Extensions for various encoded files in Asterisk
  • wav:
  • pcm:
  • gsm:

Packetization

Various clients support variable sample periods / packetization. Asterisk 1.2 and earlier only supports 20ms packetization in RTP-based protocols like SIP and MGCP, so you should configure your client to use this. However, iLBC with its 30 ms packets also works with Asterisk 1.2. 1.4 and later include support for variable packetization, either settable in the config or set automatically according to the SDP.

See also:

Created by flavour, Last modification by rkarlsba on Sun 28 of Jan, 2007 [12:09 UTC]

Comments

Comments Filter

variables for codecs?

by Dirk-Michael Brosig on Thursday 14 of December, 2006 [09:55:37 UTC]
If there variables for the used audio and video codecs for use in extension.conf? P.e. ...

exten => 100,n,GotoIf($"${CODEC_VIDEO}" = "h.264"?200:300) // play h.264 video otherwise play h.263
exten => 100,200,Playback(video264)
exten => 100,300,Playback(video263)

Packetization

by Sergey Basmanov on Wednesday 16 of August, 2006 [11:07:14 UTC]
Beta version of packetization code is available here: http://bugs.digium.com/view.php?id=5162
With this patch, packetization can be set for each codec for SIP peer/user, as well as match to remote packetization values if present in SDP.

What CODEC am I using?

by omar on Monday 24 of October, 2005 [15:27:35 UTC]
From my voice provider I have to choose a codec g.726, and I need to know what g.726 codec I am using.
How can I know if I am using
g.726 - 40
g.726 - 32
g.726 - 16
etc.
or ASTERISK only support g.726 - 32

Re: Quality Voice

by chammoud on Friday 15 of April, 2005 [09:54:23 UTC]
it could be due to many factors. Internet bandwidth one of them. if you are using iax test different codecs and see how that cahnge the quality of the sound. also in iax.conf try a call with jitterbuffer=yes and no or comment it. whaen you make a call perform this command: iax2 show netstats and write down your results.
be a member os asterisk-lists and post your problem with much more inputs so people can help you

Quality Voice

by hameds on Tuesday 08 of March, 2005 [14:21:31 UTC]
(:razz:)

Hi... i need your help .... the quality voice from my asterisk id very low how to change this!!!

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver